With the explosion in Internet access and usage as well as the wide spread usage of local area networks (LANs) and wide area networks (WANs) attempts have been made to use these forms of digital communications technology for voice communications known as voice over Internet protocol (VoIP). The primary benefit in using VoIP over the public switched telephone network (PSTN) is the substantial savings that can be realized in local and particularly long distance telephone bills. In addition, businesses operating LANs and/or WANs have found that they can realize a significant cost savings by utilizing their LAN or WAN instead of a more traditional private branch exchange (PBX) system. However, the quality of VoIP has proven to be inferior to that seen in the PSTN. All too often the voice transmission using VoIP is heard with gaps, delays, and noise interspersed in the conversation. At times when the network is experiencing high traffic conditions, this distortion can be so severe that a normal conversation becomes almost impossible.
Unfortunately, the cause of these voice problems is found in the very foundation of how packet switched Internet protocol (IP) networks, such as LANs, WANs and the Internet, transmit and receive information as compared to a PSTN. The PSTN was designed for optimal voice quality and provides users with dedicated, end-to-end circuit connections for the duration of each call. Circuits are reserved between the originating switch, tandem switches (if any), and the terminating switch based on the called party number. Therefore, the user in the PSTN has a dedicated communications line completely at their disposal for the duration of the call even when no information is being transmitted.
Unlike the circuit-switched PSTN, packet-switched IP networks provide virtual circuit connections between users. Bandwidth is shared for improved utilization of network capacity leading to lower costs for network users. Thus, packet switched IP networks were designed for the efficient transmission of computer data and not the transmission of sounds as they are generated. In packet switched IP networks, large volumes of data being transmitted are first divided into packets of a fixed or more often a variable length. The assembly of these packets entails the creation of a header having at least a packet sequence number, a source address, a destination address and packet size, contained therein. The individual packets containing the header and data are then transmitted, usually to a gateway server, and then to routers in the case of the Internet. The routers take the data and then transmit it to routers located closer to the ultimate destination, taking into consideration traffic conditions, until the final destination is reached. The number of packets assembled and transmitted is directly dependent on the volume of data being transmitted. Also, the route each packet takes to the destination may vary from packet to packet. Further, the number of routers a particular packet must pass through may vary based on the route taken and traffic conditions.
Therefore, since each data packet may take a different route to the destination, the sequence of arrival for each packet may not match that of transmission. Further, in the transmission process, often a data packet is lost due to corruption of the header information. When dealing with computer related data, the out of sequence arrival of packets and the loss of a packet is not a problem since the receiving computer can either wait for arrival of the packet or request retransmission of the packet if it does not arrive in a predetermined time period or when the data received is corrupted. Even in the case where a user is waiting for the downloading of graphic information, a short delay or interruption in transmission of the image is not often considered a significant problem. Even the complete loss of graphic data is not a problem since it can be retransmitted and ultimately cause only another delay. However, when conducting a telephone conversation or listening to music, even a very brief delay or interruption of reception is so disconcerting to the listener that it is completely unacceptable. Further, when the traffic on a packet switched IP network increases the more frequent these delays, interruptions and lost sounds become. Again, as traffic increases on a packet switched IP network the quality of the transmission can degrade to the point where normal voice communications is not possible.
Attempts to alleviate the delay have employed faster modems and communications lines. Further, attempts have been made to prioritize packets containing voice data so that they are serviced ahead of other types of data by routers. However, these efforts have had limited success and have not solved the fundamental problem that in packet switched IP networks as traffic increases quality will eventually suffer. No matter how fast the hardware or how sophisticated software, this fundamental problem generated by excessive volume will persist. Further, this drop off or decline in voice quality seen in a packet switched IP network is not necessarily a sloping curve. Instead, with the addition of a handful of the new active callers in a packet switched IP network the quality of transmissions may suddenly drop-off to an unacceptable level for all callers.
Therefore, what is needed is a computer program and method that can use the packet switched IP networks for voice transmission, thereby realizing a cost savings over PSTN, and mitigate or eliminate the effects heard by a listener caused by lost or delayed packets containing voice and sound data which can become acute when call volume increases. This computer program and method should be able to determine VoIP call volume and prevent or block the entry of additional callers when either a predetermined number of callers is active or when monitoring of the network indicates that voice quality may be about to drop off.